Merge "hal: Fix incorrect wsa count update during platform_init"
diff --git a/configs/msm8909/msm8909.mk b/configs/msm8909/msm8909.mk
index 9a32dee..f9ce14c 100755
--- a/configs/msm8909/msm8909.mk
+++ b/configs/msm8909/msm8909.mk
@@ -135,6 +135,10 @@
PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.dolby.ds2.enabled=true
+#Disable split a2dp
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.enable.splita2dp=false
+
PRODUCT_PACKAGES += \
android.hardware.audio@2.0-service \
android.hardware.audio@2.0-impl \
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index 6200705..0cf380c 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -228,6 +228,10 @@
PRODUCT_PROPERTY_OVERRIDES += \
ro.af.client_heap_size_kbyte=7168
+#Disable split a2dp
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.enable.splita2dp=false
+
# for HIDL related packages
PRODUCT_PACKAGES += \
android.hardware.audio@2.0-service \
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index 068317d..83baae5 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -227,6 +227,10 @@
PRODUCT_PROPERTY_OVERRIDES += \
ro.af.client_heap_size_kbyte=7168
+#Disable split a2dp
+PRODUCT_PROPERTY_OVERRIDES += \
+persist.vendor.bt.enable.splita2dp=false
+
# for HIDL related packages
PRODUCT_PACKAGES += \
android.hardware.audio@2.0-service \
diff --git a/hal/Makefile.am b/hal/Makefile.am
index 95224d0..46882f1 100644
--- a/hal/Makefile.am
+++ b/hal/Makefile.am
@@ -4,6 +4,7 @@
-I $(top_srcdir)/hal \
-I $(top_srcdir)/hal/audio_extn \
-I $(top_srcdir)/hal/voice_extn \
+ -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/audio-kernel \
-I $(top_srcdir)/hal/${TARGET_PLATFORM}
c_sources = audio_hw.c \
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index af8cc89..8766c0e 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, 2016-2017 The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013, 2016-2018 The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -51,6 +51,7 @@
#define ABS_SUB(A, B) (((A) > (B)) ? ((A) - (B)):((B) - (A)))
#define SAMPLE_RATE_8000 8000
#define SAMPLE_RATE_11025 11025
+#define SAMPLE_RATE_192000 192000
// Supported sample rates for USB
static uint32_t supported_sample_rates[] =
{384000, 352800, 192000, 176400, 96000, 88200, 64000, 48000, 44100, 32000, 22050, 16000, 11025, 8000};
@@ -280,6 +281,10 @@
for (i = 0; i < MAX_SAMPLE_RATE_SIZE; i++) {
if (supported_sample_rates[i] >= min_sr &&
supported_sample_rates[i] <= max_sr) {
+ // FIXME: we don't support >192KHz in recording path for now
+ if ((supported_sample_rates[i] > SAMPLE_RATE_192000) &&
+ (type == USB_CAPTURE))
+ continue;
config->rates[sr_size++] = supported_sample_rates[i];
supported_sample_rates_mask[type] |= (1<<i);
ALOGI_IF(usb_audio_debug_enable,
@@ -290,6 +295,12 @@
} else {
do {
sr = (uint32_t)atoi(next_sr_string);
+ // FIXME: we don't support >192KHz in recording path for now
+ if ((sr > SAMPLE_RATE_192000) && (type == USB_CAPTURE)) {
+ next_sr_string = strtok_r(NULL, " ,.-", &temp_ptr);
+ continue;
+ }
+
for (i = 0; i < MAX_SAMPLE_RATE_SIZE; i++) {
if (supported_sample_rates[i] == sr) {
ALOGI_IF(usb_audio_debug_enable,
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index e03fdc8..d4c6351 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014-2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2014 The Android Open Source Project
@@ -869,6 +869,25 @@
}
}
+void audio_extn_btsco_get_sample_rate(int snd_device, int *sample_rate)
+{
+ switch (snd_device) {
+ case SND_DEVICE_OUT_BT_SCO:
+ case SND_DEVICE_IN_BT_SCO_MIC:
+ case SND_DEVICE_IN_BT_SCO_MIC_NREC:
+ *sample_rate = 8000;
+ break;
+ case SND_DEVICE_OUT_BT_SCO_WB:
+ case SND_DEVICE_IN_BT_SCO_MIC_WB:
+ case SND_DEVICE_IN_BT_SCO_MIC_WB_NREC:
+ *sample_rate = 16000;
+ break;
+ default:
+ ALOGD("%s:Not a BT SCO device, need not update sampling rate\n", __func__);
+ break;
+ }
+}
+
static int send_app_type_cfg_for_device(struct audio_device *adev,
struct audio_usecase *usecase,
int split_snd_device)
@@ -935,8 +954,13 @@
goto exit_send_app_type_cfg;
}
snd_device = platform_get_spkr_prot_snd_device(snd_device);
-
- acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
+ if (voice_is_in_call_rec_stream(usecase->stream.in) && usecase->type == PCM_CAPTURE) {
+ snd_device_t voice_device = voice_get_incall_rec_snd_device(usecase->in_snd_device);
+ acdb_dev_id = platform_get_snd_device_acdb_id(voice_device);
+ ALOGV("acdb id for voice call use case %d", acdb_dev_id);
+ } else {
+ acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
+ }
if (acdb_dev_id <= 0) {
ALOGE("%s: Couldn't get the acdb dev id", __func__);
rc = -EINVAL;
@@ -992,6 +1016,7 @@
ALOGI("%s using %d sample rate rate for A2DP CoPP",
__func__, usecase->stream.out->app_type_cfg.sample_rate);
}
+ audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.out->app_type_cfg.sample_rate);
sample_rate = usecase->stream.out->app_type_cfg.sample_rate;
/* Interactive streams are supported with only direct app type id.
@@ -1034,6 +1059,11 @@
app_type_cfg[len++] = acdb_dev_id;
if (usecase->id == USECASE_AUDIO_RECORD_VOIP)
usecase->stream.in->app_type_cfg.sample_rate = usecase->stream.in->sample_rate;
+ if (voice_is_in_call_rec_stream(usecase->stream.in)) {
+ audio_extn_btsco_get_sample_rate(usecase->in_snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
+ } else {
+ audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.in->app_type_cfg.sample_rate);
+ }
sample_rate = usecase->stream.in->app_type_cfg.sample_rate;
app_type_cfg[len++] = sample_rate;
if (snd_device_be_idx > 0)
@@ -1067,6 +1097,7 @@
{
int i, num_devices = 0;
snd_device_t new_snd_devices[SND_DEVICE_OUT_END] = {0};
+ snd_device_t in_snd_device = usecase->in_snd_device;
int rc = 0;
switch (usecase->type) {
@@ -1085,11 +1116,14 @@
case PCM_CAPTURE:
ALOGD("%s: usecase->in_snd_device %s",
__func__, platform_get_snd_device_name(usecase->in_snd_device));
+ if (voice_is_in_call_rec_stream(usecase->stream.in)) {
+ in_snd_device = voice_get_incall_rec_backend_device(usecase->stream.in);
+ }
/* check for in combo device */
if (platform_split_snd_device(adev->platform,
- usecase->in_snd_device,
+ in_snd_device,
&num_devices, new_snd_devices)) {
- new_snd_devices[0] = usecase->in_snd_device;
+ new_snd_devices[0] = in_snd_device;
num_devices = 1;
}
break;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9ebe142..c7186d9 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -90,6 +90,7 @@
#define DIRECT_PCM_NUM_FRAGMENTS 2
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
#define VOIP_PLAYBACK_VOLUME_MAX 0x2000
+#define PCM_PLAYBACK_VOLUME_MAX 0x2000
#define DSD_VOLUME_MIN_DB (-110)
#define PROXY_OPEN_RETRY_COUNT 100
@@ -443,6 +444,7 @@
static int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore);
static int out_set_compr_volume(struct audio_stream_out *stream, float left, float right);
static int out_set_voip_volume(struct audio_stream_out *stream, float left, float right);
+static int out_set_pcm_volume(struct audio_stream_out *stream, float left, float right);
static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
int flags __unused)
@@ -1423,6 +1425,11 @@
usecase->out_snd_device,
platform_get_input_snd_device(adev->platform, uc_info->devices));
enable_audio_route(adev, usecase);
+ if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
+ out_set_voip_volume(&usecase->stream.out->stream,
+ usecase->stream.out->volume_l,
+ usecase->stream.out->volume_r);
+ }
if (usecase->id == USECASE_AUDIO_PLAYBACK_FM) {
struct str_parms *parms = str_parms_create_str("fm_restore_volume=1");
if (parms)
@@ -3094,6 +3101,8 @@
// apply volume for voip playback after path is set up
if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP)
out_set_voip_volume(&out->stream, out->volume_l, out->volume_r);
+ else if ((out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY || out->usecase == USECASE_AUDIO_PLAYBACK_DEEP_BUFFER))
+ out_set_pcm_volume(&out->stream, out->volume_l, out->volume_r);
} else {
platform_set_stream_channel_map(adev->platform, out->channel_mask,
out->pcm_device_id, &out->channel_map_param.channel_map[0]);
@@ -4150,6 +4159,33 @@
return 0;
}
+static int out_set_pcm_volume(struct audio_stream_out *stream, float left,
+ float right)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ /* Volume control for pcm playback */
+ if (left != right) {
+ return -EINVAL;
+ } else {
+ char mixer_ctl_name [128];
+ struct audio_device *adev = out->dev;
+ struct mixer_ctl *ctl;
+ int pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "Playback %d Volume", pcm_device_id);
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s : Could not get ctl for mixer cmd - %s", __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+ int volume = (int) (left * PCM_PLAYBACK_VOLUME_MAX);
+ mixer_ctl_set_value(ctl, 0, volume);
+
+ ALOGV("%s : Pcm set volume value %d", __func__, volume);
+
+ return 0;
+ }
+}
+
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
@@ -4199,6 +4235,13 @@
out->volume_l = left;
out->volume_r = right;
return ret;
+ } else {
+ /* Volume control for pcm playback */
+ if (!out->standby)
+ ret = out_set_pcm_volume(stream, left, right);
+ out->volume_l = left;
+ out->volume_r = right;
+ return ret;
}
return -ENOSYS;
@@ -4651,43 +4694,27 @@
clock_gettime(CLOCK_MONOTONIC, timestamp);
} else {
if (out->pcm) {
- int64_t signed_frames = -1;
- // XXX it might be better to identify these
- // as realtime usecases?
- if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP ||
- out->usecase == USECASE_AUDIO_PLAYBACK_ULL) {
- unsigned int hw_ptr;
- if (pcm_mmap_get_hw_ptr(out->pcm, &hw_ptr, timestamp) == 0) {
- signed_frames = hw_ptr;
- }
- ALOGV("%s frames %lld", __func__, (long long)signed_frames);
- } else {
- unsigned int avail;
- if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
- size_t kernel_buffer_size =
- out->config.period_size * out->config.period_count;
- signed_frames =
- out->written - kernel_buffer_size + avail;
- }
- }
-
- // This adjustment accounts for buffering after app processor.
- // It is based on estimated DSP latency per use case, rather than exact.
- signed_frames -=
- (platform_render_latency(out->usecase) *
- out->sample_rate / 1000000LL);
-
- // Adjustment accounts for A2dp encoder latency with non offload usecases
- // Note: Encoder latency is returned in ms, while platform_render_latency in us.
- if (AUDIO_DEVICE_OUT_ALL_A2DP & out->devices) {
+ unsigned int avail;
+ if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
+ size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
+ int64_t signed_frames = out->written - kernel_buffer_size + avail;
+ // This adjustment accounts for buffering after app processor.
+ // It is based on estimated DSP latency per use case, rather than exact.
signed_frames -=
- (audio_extn_a2dp_get_encoder_latency() * out->sample_rate / 1000);
- }
+ (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
- // It would be unusual for this value to be negative, but check just in case ...
- if (signed_frames >= 0) {
- *frames = signed_frames;
- ret = 0;
+ // Adjustment accounts for A2dp encoder latency with non offload usecases
+ // Note: Encoder latency is returned in ms, while platform_render_latency in us.
+ if (AUDIO_DEVICE_OUT_ALL_A2DP & out->devices) {
+ signed_frames -=
+ (audio_extn_a2dp_get_encoder_latency() * out->sample_rate / 1000);
+ }
+
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ ret = 0;
+ }
}
} else if (out->card_status == CARD_STATUS_OFFLINE) {
*frames = out->written;
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index b0c24ac..2e824b0 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -555,6 +555,9 @@
[SND_DEVICE_IN_EC_REF_LOOPBACK_MONO] = "ec-ref-loopback-mono",
[SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO] = "ec-ref-loopback-stereo",
[SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = "quad-mic",
+ [SND_DEVICE_IN_INCALL_REC_RX] = "incall-rec-rx",
+ [SND_DEVICE_IN_INCALL_REC_TX] = "incall-rec-tx",
+ [SND_DEVICE_IN_INCALL_REC_RX_TX] = "incall-rec-rx-tx",
};
// Platform specific backend bit width table
@@ -868,6 +871,9 @@
{TO_NAME_INDEX(SND_DEVICE_IN_EC_REF_LOOPBACK_MONO)},
{TO_NAME_INDEX(SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_GENERIC_QMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_INCALL_REC_RX)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_INCALL_REC_TX)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_INCALL_REC_RX_TX)},
};
static char * backend_tag_table[SND_DEVICE_MAX] = {0};
@@ -1682,6 +1688,8 @@
hw_interface_table[SND_DEVICE_IN_UNPROCESSED_QUAD_MIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = strdup("SLIMBUS_0_TX");
+ hw_interface_table[SND_DEVICE_IN_INCALL_REC_RX] = strdup("INCALL_RECORD_RX");
+ hw_interface_table[SND_DEVICE_IN_INCALL_REC_TX] = strdup("INCALL_RECORD_TX");
my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
/*remove ALAC & APE from DSP decoder list based on software decoder availability*/
@@ -3795,6 +3803,11 @@
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
ret = 0;
+ } else if (SND_DEVICE_IN_INCALL_REC_RX_TX == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_IN_INCALL_REC_RX;
+ new_snd_devices[1] = SND_DEVICE_IN_INCALL_REC_TX;
+ ret = 0;
}
ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index aeefcc2..c610e4c 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2018, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -229,6 +229,9 @@
SND_DEVICE_IN_EC_REF_LOOPBACK_MONO,
SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO,
SND_DEVICE_IN_HANDSET_GENERIC_QMIC,
+ SND_DEVICE_IN_INCALL_REC_RX,
+ SND_DEVICE_IN_INCALL_REC_TX,
+ SND_DEVICE_IN_INCALL_REC_RX_TX,
SND_DEVICE_IN_END,
SND_DEVICE_MAX = SND_DEVICE_IN_END,
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 475e438..e370641 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -539,6 +539,9 @@
[SND_DEVICE_IN_UNPROCESSED_QUAD_MIC] = "unprocessed-quad-mic",
[SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = "unprocessed-headset-mic",
[SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = "quad-mic",
+ [SND_DEVICE_IN_INCALL_REC_RX] = "incall-rec-rx",
+ [SND_DEVICE_IN_INCALL_REC_TX] = "incall-rec-tx",
+ [SND_DEVICE_IN_INCALL_REC_RX_TX] = "incall-rec-rx-tx",
};
// Platform specific backend bit width table
@@ -849,6 +852,9 @@
{TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_QUAD_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_GENERIC_QMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_INCALL_REC_RX)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_INCALL_REC_TX)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_INCALL_REC_RX_TX)},
};
static char * backend_tag_table[SND_DEVICE_MAX] = {0};
@@ -1539,6 +1545,8 @@
hw_interface_table[SND_DEVICE_IN_UNPROCESSED_QUAD_MIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_HANDSET_GENERIC_QMIC] = strdup("SLIMBUS_0_TX");
+ hw_interface_table[SND_DEVICE_IN_INCALL_REC_RX] = strdup("INCALL_RECORD_RX");
+ hw_interface_table[SND_DEVICE_IN_INCALL_REC_TX] = strdup("INCALL_RECORD_TX");
my_data->max_mic_count = PLATFORM_DEFAULT_MIC_COUNT;
@@ -3604,6 +3612,11 @@
new_snd_devices[0] = SND_DEVICE_OUT_SPEAKER;
new_snd_devices[1] = SND_DEVICE_OUT_BT_A2DP;
ret = 0;
+ } else if (SND_DEVICE_IN_INCALL_REC_RX_TX == snd_device) {
+ *num_devices = 2;
+ new_snd_devices[0] = SND_DEVICE_IN_INCALL_REC_RX;
+ new_snd_devices[1] = SND_DEVICE_IN_INCALL_REC_TX;
+ ret = 0;
}
ALOGD("%s: snd_device(%d) num devices(%d) new_snd_devices(%d)", __func__,
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 8151ea6..04f25de 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -234,6 +234,9 @@
SND_DEVICE_IN_EC_REF_LOOPBACK_MONO,
SND_DEVICE_IN_EC_REF_LOOPBACK_STEREO,
SND_DEVICE_IN_HANDSET_GENERIC_QMIC,
+ SND_DEVICE_IN_INCALL_REC_RX,
+ SND_DEVICE_IN_INCALL_REC_TX,
+ SND_DEVICE_IN_INCALL_REC_RX_TX,
SND_DEVICE_IN_END,
SND_DEVICE_MAX = SND_DEVICE_IN_END,
diff --git a/hal/voice.c b/hal/voice.c
index f9e3562..0a48a05 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2018, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -436,6 +436,27 @@
return ret;
}
+snd_device_t voice_get_incall_rec_backend_device(struct stream_in *in)
+{
+ snd_device_t incall_record_device = {0};
+
+ switch(in->source) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ incall_record_device = SND_DEVICE_IN_INCALL_REC_TX;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ incall_record_device = SND_DEVICE_IN_INCALL_REC_RX;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ incall_record_device = SND_DEVICE_IN_INCALL_REC_RX_TX;
+ break;
+ default:
+ ALOGI("Invalid source %d", in->source);
+ }
+
+ return incall_record_device;
+}
+
snd_device_t voice_get_incall_rec_snd_device(snd_device_t in_snd_device)
{
snd_device_t incall_record_device = in_snd_device;
diff --git a/hal/voice.h b/hal/voice.h
index 3ae42a8..ca46e24 100644
--- a/hal/voice.h
+++ b/hal/voice.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2016,2018, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -101,4 +101,5 @@
snd_device_t out_snd_device,
bool enable);
bool voice_is_call_state_active(struct audio_device *adev);
+snd_device_t voice_get_incall_rec_backend_device(struct stream_in *in);
#endif //VOICE_H
diff --git a/mm-audio/aenc-aac/qdsp6/Makefile.am b/mm-audio/aenc-aac/qdsp6/Makefile.am
index 08c9bee..67f0bf1 100644
--- a/mm-audio/aenc-aac/qdsp6/Makefile.am
+++ b/mm-audio/aenc-aac/qdsp6/Makefile.am
@@ -17,6 +17,7 @@
AM_CPPFLAGS += -DNDEBUG
AM_CPPFLAGS += -Iinc
AM_CPPFLAGS += -I ${WORKSPACE}/hardware/qcom/media/mm-core/inc/
+AM_CPPFLAGS += -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/audio-kernel
c_sources =src/omx_aac_aenc.cpp
c_sources +=src/aenc_svr.c
diff --git a/mm-audio/aenc-amrnb/qdsp6/Makefile.am b/mm-audio/aenc-amrnb/qdsp6/Makefile.am
index 8becd07..1b7ed7c 100644
--- a/mm-audio/aenc-amrnb/qdsp6/Makefile.am
+++ b/mm-audio/aenc-amrnb/qdsp6/Makefile.am
@@ -17,6 +17,7 @@
AM_CPPFLAGS += -DFEATURE_NATIVELINUX -Dstrlcpy=g_strlcpy
AM_CPPFLAGS += -DFEATURE_DSM_DUP_ITEMS
AM_CPPFLAGS += -I ${WORKSPACE}/hardware/qcom/media/mm-core/inc/
+AM_CPPFLAGS += -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/audio-kernel
AM_CPPFLAGS += -g
AM_CPPFLAGS += -DNDEBUG
diff --git a/mm-audio/aenc-g711/qdsp6/Makefile.am b/mm-audio/aenc-g711/qdsp6/Makefile.am
index 02b0d57..9f42449 100644
--- a/mm-audio/aenc-g711/qdsp6/Makefile.am
+++ b/mm-audio/aenc-g711/qdsp6/Makefile.am
@@ -11,6 +11,7 @@
-DNDEBUG \
-DAUDIOV2 \
-I inc \
+ -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/audio-kernel \
-I ${WORKSPACE}/hardware/qcom/media/mm-core/inc/
C_SOURCES = src/aenc_svr.c \
diff --git a/post_proc/Makefile.am b/post_proc/Makefile.am
index e161f8d..54602a2 100644
--- a/post_proc/Makefile.am
+++ b/post_proc/Makefile.am
@@ -1,4 +1,5 @@
AM_CFLAGS = -I ${WORKSPACE}/external/tinyalsa/include \
+ -I $(PKG_CONFIG_SYSROOT_DIR)/usr/include/audio-kernel \
-I ${WORKSPACE}/system/media/audio_effects/include
c_sources = bundle.c \
diff --git a/qahw_api/test/qahw_multi_record_test.c b/qahw_api/test/qahw_multi_record_test.c
index 2e0a396..e00ef7b 100644
--- a/qahw_api/test/qahw_multi_record_test.c
+++ b/qahw_api/test/qahw_multi_record_test.c
@@ -75,6 +75,8 @@
double record_length;
char profile[50];
char kvpairs[256];
+ bool timestamp_mode;
+ char timestamp_file_in[256];
};
struct timed_params {
@@ -261,6 +263,7 @@
char file_name[256] = "/data/rec";
int data_sz = 0, name_len = strlen(file_name);
qahw_in_buffer_t in_buf;
+ static int64_t timestamp = 1;
struct audio_config_params* params = (struct audio_config_params*) thread_param;
qahw_module_handle_t *qahw_mod_handle = params->qahw_mod_handle;
@@ -358,6 +361,17 @@
test_end();
pthread_exit(0);
}
+
+ FILE *fd_in_ts;
+ if (*(params->timestamp_file_in))
+ fd_in_ts = fopen(params->timestamp_file_in, "w+");
+ if (fd_in_ts == NULL) {
+ fprintf(log_file, "playback timestamps file open failed \n");
+ if (log_file != stdout)
+ fprintf(stdout, "playback timestamps file open failed \n");
+ test_end();
+ pthread_exit(0);
+ }
int bps = 16;
switch(params->config.format) {
@@ -411,8 +425,12 @@
in_buf.buffer = buffer;
in_buf.bytes = buffer_size;
+ if (params->timestamp_mode)
+ in_buf.timestamp = ×tamp;
bytes_read = qahw_in_read(in_handle, &in_buf);
+ if (params->timestamp_mode)
+ fprintf(fd_in_ts, "timestamp:%lu\n", timestamp);
if (kpi_mode) {
if (count == 0) {
ret = clock_gettime(CLOCK_REALTIME, &tsColdF);
@@ -437,6 +455,7 @@
}
data_sz += buffer_size;
}
+ fclose(fd_in_ts);
/*Stopping sourcetracking thread*/
sourcetrack_done = 1;
@@ -548,6 +567,7 @@
thread_param->source = 1;
thread_param->record_length = 8 /*sec*/;
thread_param->record_delay = 0 /*sec*/;
+ thread_param->timestamp_mode = false;
thread_param->handle = 0x99A - rec_session;
}
@@ -568,6 +588,7 @@
printf(" -D --recording-delay <in seconds> - Delay in seconds after which recording should be started\n\n");
printf(" -l --log-file <FILEPATH> - File path for debug msg, to print\n");
printf(" on console use stdout or 1 \n\n");
+ printf(" -m --timestamp-mode <FILEPATH> - Use this flag to support timestamp-mode and timestamp file path for debug msg\n");
printf(" -K --kpi-mode - Use this flag to measure latency KPIs for this recording\n\n");
printf(" -i --interactive-mode - Use this flag if prefer configuring streams using interactive mode\n");
printf(" All other flags passed would be ignore if this flag is used\n\n");
@@ -586,6 +607,8 @@
printf(" hal_rec_test -F 1 --kpi-mode -> start a recording with low latency input flag and calculate latency KPIs\n\n");
printf(" hal_rec_test -c 1 -r 16000 -t 30 -k ffvOn=true;ffv_ec_ref_ch_cnt=2 -> Enable FFV with stereo ec ref\n");
printf(" For mono channel 16kHz rate for 30seconds\n\n");
+ printf(" hal_rec_test -d 2 -f 1 -r 44100 -c 2 -t 8 -D 2 -m <FILEPATH> -F 2147483648 --> enable timestamp mode and\n");
+ printf(" print timestamp debug msg in specified FILEPATH\n");
}
static void qti_audio_server_death_notify_cb(void *ctxt __unused) {
@@ -622,6 +645,7 @@
{"recording-time", required_argument, 0, 't'},
{"recording-delay", required_argument, 0, 'D'},
{"log-file", required_argument, 0, 'l'},
+ {"timestamp-file", required_argument, 0, 'm'},
{"kpi-mode", no_argument, 0, 'K'},
{"interactive", no_argument, 0, 'i'},
{"source-tracking", no_argument, 0, 'S'},
@@ -634,7 +658,7 @@
int option_index = 0;
while ((opt = getopt_long(argc,
argv,
- "-d:f:F:r:c:s:p:t:D:l:k:KiSh",
+ "-d:f:F:r:c:s:p:t:D:l:m:k:KiSh",
long_options,
&option_index)) != -1) {
switch (opt) {
@@ -668,6 +692,10 @@
case 'l':
snprintf(log_filename, sizeof(log_filename), "%s", optarg);
break;
+ case 'm':
+ params[0].timestamp_mode = true;
+ snprintf(params[0].timestamp_file_in, sizeof(params[0].timestamp_file_in), "%s", optarg);
+ break;
case 'K':
kpi_mode = true;
break;